1. Field of the Invention
The present invention relates to a multimedia information communication system, and particularly to a method and apparatus for controlling multimedia communication so that the communication mode is automatically changed over in accordance with a load condition of a packet switching network in a system in which multimedia information communication is performed between two terminal stations through the packet switching network.
2. Description of the Related Art
Recently, TV phone systems and TV conference systems, in which terminal stations at remote locations are connected through communication lines have been developed. In video communication, the quantity of information to be transmitted is large and it is therefore desirable that a transmission bandwidth which is in accord with the information quantity can be secured. Presently, in a local area network provided in an office or the like, however, it is impossible for every user to secure necessary transmission bandwidth because of limitations in transmission capacity.
In a network to which terminal stations which may generate a large quantity of information such as video signals are connected, problems include that the bandwidth available by the network is changed greatly or that jitter is generated on delay time of information transmission in the network for reasons such as the data suddenly bursting into the network from one of the terminal stations.
Conventionally, for example, JP-A 3-270430 teaches a voice and video communication system which is provided with a function that a TV conference terminal station connected to an LAN measures the quantity of traffic to be sent out from its own side to the LAN and the current quantity of traffic on the LAN, and when both the measured traffic quantities are not less than predetermined threshold values with respect to video and voice signals, the terminal station gives a command to a voice coder-decoder (hereinafter referred to as "codec") and a video codec so as to reduce the information transmission quantity to the LAN.
Upon reception of such an information quantity reducing command, in the voice and video communication system, the video codec lowers the upper limit value of the information transmission quantity and the voice codec operates to reduce the information transmission quantity so that the increase of the end-to-end transmission delay is suppressed to thereby prevent the video and voice signals quality from deteriorating in the video and voice. In the above-mentioned technique, however, there is a problem that when the transmission delay is large even if the video and voice transmission quantity is reduced to its minimum value, it is impossible to maintain the quality of both media in the terminal station on the receiving side.
Further, JP-A 2-209043 teaches a data communication system in which an information frame transmission side terminal station judges the traffic state in a network on the basis of the receiving state of an arrival confirm frame/re-send request frame transmitted from a receive side terminal station, and when the transmission side terminal station concludes that the traffic quantity is large, the transmission side terminal station prolongs the send interval of the information frames to thereby control the traffic which flows into the network. This system however has a problem in that if this system is applied to a voice and video communication system which takes a serious view of real time property, the response operation of traffic control is slow because the traffic state is judged on the basis of the reception state of an arrival confirm frame/re-send request frame from the data receiving side.